similar to: How to stop IVR once system receives DTMF?

Hello, In our previous PBX we have an option to turn off or on outside calls with a pincode.. Like, user is able to get calls or dial local lines by default, but when he/she uses a password entrance via dtmf, he can dial long distance calls etc.And at anytime he can logoff from outside call permit.. So is it possible to do smthing like this on asterisk.. A limited profile which needs sip password

Hi, We are using MixMonitor to record the call. When the call is bridged, the latency is significant. We tried to increase the internet speed and the server RAM and processor speed and still we are having that issue. We use VoiceTrading and Gafachi's Termination minutes to make calls. As we are in US and VoiceTrading in Europe, somebody suggested to move the termination minute provider

Hi Gang, I'm testing 1.8.0 on one of my machines and this snippet "chokes" on line 7 (works fine with 1.4.30) [tb-account-balance] exten => s,1,Set(BALCOUNT=0) exten => s,n,NoOp(Verbose(acct ${digitacc} pwd ${digitpwd} )) exten => s,n(runagi),Set(TEST_RETURN="NONE") exten =>

Hi, I am pretty new to Asterisk. I am trying to make sure some human being answers the phone not the voice mail machine. How can I programmatically identify that? Here is my Sub: sub DialPhysician { my ($self, $con, $PhysicianPhone, $call_id, $conv_id) = (@_); to_log($self, "Inside Dial Physician", 2); my $DocPhone = "1".

hi, i use $AGI->wait_for_digit($timeout) to wait for the user press key 1 ,and then to do something. but how can i get the return number ? is that use $key = $AGI->wait_for_digit($timeout) and $key will be "200 result=49" if i pressed number 1? Thanks! -- Best regards, Sucan

Hello everybody!!! I really need your help guys, I am using the AGI mode in meetme application, and I want that AGI should wait for an input from the client/user i.e. a digit and then proceed, but I have used that AGI function agi->wait_for_digit(), but no use....my agi just passes, or ignores this function, where AGI should stop here and wait for the input.... .....my extension in my

Hello Han, Thanks for the response. The XML option with qemu:commandline works if the device is added during the VM creation. But I would like to hot-plug the device to the running VM. I can add the device using the command line <virsh qemu-monitor-command --hmp VM-Name device_add edu,id=edu0> But I would like to add the device using the XML file, like <virsh attach-device VM-Name

>>> Why are you using a non default port for SMB ? The defaults are 139 and 445. I not aware of the complete details but we have a dependency for the smb server to support multiple domains, so we spin up a container with samba for each of the domains and a front server routes it to the appropriate container. That is the reason for having non-default ports and binding interfaces.

Hi Team, I am exploring "edu" device in QEMU to add my own custom device. I am able to add the device using the command line. Would it be possible to add the device using XML file through libvirt? -- Regards, Bharath

>>> What are the DC's ? DC is a read-write windows active directory domain controller on Windows server 2016. >>> Why are you using NTLMv2 ? what is it required for ? The smb client here is a windows server 2016 machine part of a domain and the smb server is on Ubuntu 22.04. The communication between the client and the server uses NTLMv2 while the communication between the

On further investigation, the error that shows up in packet capture is that the DC is returning [Fault: nca_s_fault_sec_pkg_error] for the NetrLogonSamLogonEx call. There are no error logs (or any logs) regarding the netlogon call failure in the netlogon logs even after enabling debug logs in the DC. One more interesting thing is restarting the netlogon.exe service on the DC also fixes the issue

First 'winbind enum' lines, they can and do slow things down in large domains and aren't required at all, getent etc will work without them. there are some old programs that will not work without them, but when was the last time you ran 'finger' for instance ? I made this change and it makes some difference but doesn't fix the issue entirely. Earlier the auth calls used to

A couple of things possible, from 4.8.0 winbind must be running and your smb.conf is, to be blunt, rubbish. You need to set the workgroup, you need to have idmap config lines for the workgroup, the 'winbind enum' lines only slow things down and 'map untrusted to domain' has been removed. Winbind is running and the workgroup was set as well. I omitted some lines from the smb.conf

Has anyone here used T1s with RBS with asterisk? Cary Fitch

Hi Team, I am doing a small testing and I don't know if my expectation is correct or not. Pardon me if I am ignorant. I created a VM and the VM is running. In the hypervisor I have created ".img" file and attached this .img file to the VM. My expectation is that, if VM is writing files to the attached disk, then it should reflect in the .img file created in the hypervisor. But It is

[Mar 20 12:45:33] WARNING[4940]: app_queue.c:3136 try_calling: The device state of this queue member, SIP/3617001000, is still 'Not in Use' when it probably should not be! Please check UPGRADE.txt for correct configuration settings. [Cary Fitch] We are running 1.4.22 and this message popped up in console. It could be causing our Queues announcement problem, because if all members

> I have never heard of this problem before, but Stefan Metzmacher has > opened an MR: https://gitlab.com/samba-team/samba/-/merge_requests/3162 > And there is a bug report here: > https://bugzilla.samba.org/show_bug.cgi?id=15413 > Could this be your problem ? Thanks, that is the exact problem we are facing. Will it be backported to 4.15? On Tue, Jul 4, 2023 at 6:32?PM Bharath

I am not sure what the problems are and the reasons for the basic 64K modems used in VOIP are. I understand the compressed codecs that get the bandwidth down to 20-30 K. And perhaps the 64K units give much better potential audio than you would get on a normal POTS line. But, as phone circuits VOIP/SIP doesn't seem to perform as well as plane old phones. Multiple transcodings cause issues.

I am trying to get a queue to do more than just play music and hold calls. Specifically, making some "comforting" voice announcements would be nice. Below is the queues.conf file relevant portions. Member phone number is munged to protect the guilty. We shouldn't need the announcement source info, but I have been trying everything. The problem is with the member busy, we get no

Is there a way to program an FXO device to totally ignore incoming calls? I want to put an FXO on a Fax line so that 911 calls can be sent via that line, but all other activity on the line is between the Fax machine and the phone company. Perhaps munge the ring tone detect if nothing else? Cary